mirror of
https://github.com/saitohirga/WSJT-X.git
synced 2025-10-31 04:50:34 -04:00
Summary: Add first attempt at sound capture.
Currently display the trace in specjt, but does not seem to decode it. git-svn-id: svn+ssh://svn.code.sf.net/p/wsjt/wsjt/trunk@40 ab8295b8-cf94-4d9e-aec4-7959e3be5d79
This commit is contained in:
J C Dutton
parent
682145f8e8
commit
88c5bb9a6d
@ -58,7 +58,11 @@ subroutine audio_init(ndin,ndout)
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m3=SetThreadPriority(Thread2,THREAD_PRIORITY_BELOW_NORMAL)
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m4=ResumeThread(Thread2)
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#else
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call start_threads
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print*,'Audio INIT called.'
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ierr=start_threads(ndevin,ndevout,y1,y2,nmax,iwrite,iwave,nwave, &
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11025,NSPB,TRPeriod,TxOK,ndebug,Transmitting, &
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Tsec,ngo,nmode,tbuf,ibuf,ndsec)
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#endif
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return
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3
g.py
3
g.py
@ -16,7 +16,8 @@ def ftnstr(x):
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#------------------------------------------------------ filetime
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def filetime(t):
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i=t.rfind(".")
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# i=t.rfind(".")
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i=6
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t=t[:i][-6:]
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t=t[0:2]+":"+t[2:4]+":"+t[4:6]
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return t
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6
g1
6
g1
@ -1 +1,5 @@
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python f2py.py -c --quiet --opt="-O -cpp -DLinux -fno-second-underscore" init_rs.o encode_rs.o decode_rs.o -m Audio --"f77exec=/home/joe/bin/g95" --f90exec="/home/joe/bin/g95" -L//usr/lib/gcc-lib/i386-redhat-linux/3.2.2/ -lpthread -lg2c only: ftn_init ftn_quit audio_init spec getfile azdist0 astro0 : a2d.f90 abc441.f90 astro0.f90 audio_init.f90 azdist0.f90 blanker.f90 decode1.f90 decode2.f90 decode3.f90 ftn_init.f90 ftn_quit.f90 get_fname.f90 getfile.f90 horizspec.f90 hscroll.f90 i1tor4.f90 pix2d.f90 pix2d65.f90 rfile.f90 savedata.f90 spec.f90 wsjtgen.f90 runqqq.f90 wsjt1.f fsubs1.f fsubs.f astro.f astropak.f jtaudio.c ptt_linux.c igray.c wrapkarn.c start_threads.c cutil.c fivehz.f90
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G95=/usr/bin/g95
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COMPILER=//usr/lib/gcc-lib/i686-pc-linux-gnu/3.3.6/
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python f2py.py -c --quiet --opt="-O -cpp -DLinux -fno-second-underscore" init_rs.o encode_rs.o decode_rs.o -m Audio --f77exec=$G95 --f90exec=$G95 -L$COMPILER -lpthread -lg2c -lasound only: ftn_init ftn_quit audio_init spec getfile azdist0 astro0 : a2d.f90 abc441.f90 astro0.f90 audio_init.f90 azdist0.f90 blanker.f90 decode1.f90 decode2.f90 decode3.f90 ftn_init.f90 ftn_quit.f90 get_fname.f90 getfile.f90 horizspec.f90 hscroll.f90 i1tor4.f90 pix2d.f90 pix2d65.f90 rfile.f90 savedata.f90 spec.f90 wsjtgen.f90 runqqq.f90 wsjt1.f fsubs1.f fsubs.f astro.f astropak.f jtaudio.c ptt_linux.c igray.c wrapkarn.c start_threads.c cutil.c fivehz.f90
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419
start_threads.c
419
start_threads.c
@ -1,15 +1,428 @@
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#include <stdio.h>
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#include <stdlib.h>
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#include <pthread.h>
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#include <alsa/asoundlib.h>
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#include <inttypes.h>
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#if 0
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#define ALSA_LOG
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#define ALSA_LOG_BUFFERS
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#endif
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#define BUFFER_TIME 2000*1000
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typedef struct alsa_driver_s {
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snd_pcm_t *audio_fd;
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int capabilities;
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int open_mode;
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int has_pause_resume;
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int is_paused;
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int32_t output_sample_rate, input_sample_rate;
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double sample_rate_factor;
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uint32_t num_channels;
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uint32_t bits_per_sample;
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uint32_t bytes_per_frame;
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uint32_t bytes_in_buffer; /* number of bytes writen to audio hardware */
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int16_t *app_buffer_y1;
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int16_t *app_buffer_y2;
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int *app_buffer_offset;
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int app_buffer_length;
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snd_pcm_uframes_t buffer_size;
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snd_pcm_uframes_t period_size;
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int32_t mmap;
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} alsa_driver_t;
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alsa_driver_t alsa_driver_playback;
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alsa_driver_t alsa_driver_capture;
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void *alsa_buffers[2];
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static snd_output_t *jcd_out;
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/*
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* open the audio device for writing to
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*/
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static int ao_alsa_open(alsa_driver_t *this_gen, int32_t *input_rate, snd_pcm_stream_t direction ) {
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alsa_driver_t *this = (alsa_driver_t *) this_gen;
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char *pcm_device;
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snd_pcm_hw_params_t *params;
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snd_pcm_sw_params_t *swparams;
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snd_pcm_access_mask_t *mask;
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snd_pcm_uframes_t period_size_min;
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snd_pcm_uframes_t period_size_max;
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snd_pcm_uframes_t buffer_size_min;
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snd_pcm_uframes_t buffer_size_max;
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snd_pcm_format_t format;
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uint32_t buffer_time=BUFFER_TIME;
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snd_pcm_uframes_t buffer_time_to_size;
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int err, dir;
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int open_mode=1; /* NONBLOCK */
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/* int open_mode=0; BLOCK */
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int32_t rate=*input_rate;
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this->input_sample_rate=*input_rate;
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snd_pcm_hw_params_alloca(¶ms);
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snd_pcm_sw_params_alloca(&swparams);
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err = snd_output_stdio_attach(&jcd_out, stdout, 0);
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this->num_channels = 2;
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pcm_device="default";
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#ifdef ALSA_LOG
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printf("audio_alsa_out: Audio Device name = %s\n",pcm_device);
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printf("audio_alsa_out: Number of channels = %d\n",this->num_channels);
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#endif
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if (this->audio_fd) {
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printf("audio_alsa_out:Already open...WHY!");
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snd_pcm_close (this->audio_fd);
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this->audio_fd = NULL;
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}
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this->bytes_in_buffer = 0;
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/*
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* open audio device
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*/
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err=snd_pcm_open(&this->audio_fd, pcm_device, direction, open_mode);
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if(err <0 ) {
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printf ("audio_alsa_out: snd_pcm_open() of %s failed: %s\n", pcm_device, snd_strerror(err));
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printf ("audio_alsa_out: >>> check if another program already uses PCM <<<\n");
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return 0;
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}
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/* printf ("audio_alsa_out: snd_pcm_open() opened %s\n", pcm_device); */
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/* We wanted non blocking open but now put it back to normal */
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//snd_pcm_nonblock(this->audio_fd, 0);
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snd_pcm_nonblock(this->audio_fd, 1);
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/*
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* configure audio device
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*/
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err = snd_pcm_hw_params_any(this->audio_fd, params);
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if (err < 0) {
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printf ("audio_alsa_out: broken configuration for this PCM: no configurations available: %s\n"),
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snd_strerror(err);
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goto close;
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}
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/* set interleaved access */
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if (this->mmap != 0) {
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mask = alloca(snd_pcm_access_mask_sizeof());
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snd_pcm_access_mask_none(mask);
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snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_INTERLEAVED);
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snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
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snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_COMPLEX);
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err = snd_pcm_hw_params_set_access_mask(this->audio_fd, params, mask);
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if (err < 0) {
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printf ( "audio_alsa_out: mmap not availiable, falling back to compatiblity mode\n");
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this->mmap=0;
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err = snd_pcm_hw_params_set_access(this->audio_fd, params,
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SND_PCM_ACCESS_RW_NONINTERLEAVED);
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}
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} else {
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err = snd_pcm_hw_params_set_access(this->audio_fd, params,
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SND_PCM_ACCESS_RW_NONINTERLEAVED);
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}
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if (err < 0) {
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printf ( "audio_alsa_out: access type not available: %s\n", snd_strerror(err));
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goto close;
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}
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/* set the sample format S16 */
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/* ALSA automatically appends _LE or _BE depending on the CPU */
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format = SND_PCM_FORMAT_S16;
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err = snd_pcm_hw_params_set_format(this->audio_fd, params, format );
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if (err < 0) {
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printf ( "audio_alsa_out: sample format non available: %s\n", snd_strerror(err));
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goto close;
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}
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/* set the number of channels */
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err = snd_pcm_hw_params_set_channels(this->audio_fd, params, this->num_channels);
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if (err < 0) {
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printf ( "audio_alsa_out: Cannot set number of channels to %d (err=%d:%s)\n",
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this->num_channels, err, snd_strerror(err));
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goto close;
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}
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#if SND_LIB_VERSION >= 0x010009
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/* Restrict a configuration space to contain only real hardware rates */
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err = snd_pcm_hw_params_set_rate_resample(this->audio_fd, params, 0);
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#endif
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/* set the stream rate [Hz] */
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dir=0;
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err = snd_pcm_hw_params_set_rate_near(this->audio_fd, params, &rate, &dir);
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if (err < 0) {
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printf ( "audio_alsa_out: rate not available: %s\n", snd_strerror(err));
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goto close;
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}
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this->output_sample_rate = (uint32_t)rate;
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if (this->input_sample_rate != this->output_sample_rate) {
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printf ( "audio_alsa_out: audio rate : %d requested, %d provided by device/sec\n",
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this->input_sample_rate, this->output_sample_rate);
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}
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buffer_time_to_size = ( (uint64_t)buffer_time * rate) / 1000000;
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err = snd_pcm_hw_params_get_buffer_size_min(params, &buffer_size_min);
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err = snd_pcm_hw_params_get_buffer_size_max(params, &buffer_size_max);
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dir=0;
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err = snd_pcm_hw_params_get_period_size_min(params, &period_size_min,&dir);
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dir=0;
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err = snd_pcm_hw_params_get_period_size_max(params, &period_size_max,&dir);
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#ifdef ALSA_LOG_BUFFERS
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printf("Buffer size range from %lu to %lu\n",buffer_size_min, buffer_size_max);
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printf("Period size range from %lu to %lu\n",period_size_min, period_size_max);
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printf("Buffer time size %lu\n",buffer_time_to_size);
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#endif
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this->buffer_size = buffer_time_to_size;
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if (buffer_size_max < this->buffer_size) this->buffer_size = buffer_size_max;
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if (buffer_size_min > this->buffer_size) this->buffer_size = buffer_size_min;
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this->period_size=this->buffer_size/8;
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this->buffer_size = this->period_size*8;
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#ifdef ALSA_LOG_BUFFERS
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printf("To choose buffer_size = %ld\n",this->buffer_size);
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printf("To choose period_size = %ld\n",this->period_size);
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#endif
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#if 0
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/* Set period to buffer size ratios at 8 periods to 1 buffer */
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dir=-1;
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periods=8;
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err = snd_pcm_hw_params_set_periods_near(this->audio_fd, params, &periods ,&dir);
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if (err < 0) {
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xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
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"audio_alsa_out: unable to set any periods: %s\n", snd_strerror(err));
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goto close;
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}
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/* set the ring-buffer time [us] (large enough for x us|y samples ...) */
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dir=0;
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err = snd_pcm_hw_params_set_buffer_time_near(this->audio_fd, params, &buffer_time, &dir);
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if (err < 0) {
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xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
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"audio_alsa_out: buffer time not available: %s\n", snd_strerror(err));
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goto close;
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}
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#endif
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#if 1
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/* set the period time [us] (interrupt every x us|y samples ...) */
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dir=0;
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err = snd_pcm_hw_params_set_period_size_near(this->audio_fd, params, &(this->period_size), &dir);
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if (err < 0) {
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printf ( "audio_alsa_out: period time not available: %s\n", snd_strerror(err));
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goto close;
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}
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#endif
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dir=0;
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err = snd_pcm_hw_params_get_period_size(params, &(this->period_size), &dir);
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dir=0;
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err = snd_pcm_hw_params_set_buffer_size_near(this->audio_fd, params, &(this->buffer_size));
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if (err < 0) {
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printf ( "audio_alsa_out: buffer time not available: %s\n", snd_strerror(err));
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goto close;
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}
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err = snd_pcm_hw_params_get_buffer_size(params, &(this->buffer_size));
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#ifdef ALSA_LOG_BUFFERS
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printf("was set period_size = %ld\n",this->period_size);
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printf("was set buffer_size = %ld\n",this->buffer_size);
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#endif
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if (2*this->period_size > this->buffer_size) {
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printf ( "audio_alsa_out: buffer to small, could not use\n");
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goto close;
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}
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/* write the parameters to device */
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err = snd_pcm_hw_params(this->audio_fd, params);
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if (err < 0) {
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printf ( "audio_alsa_out: pcm hw_params failed: %s\n", snd_strerror(err));
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goto close;
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}
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/* Check for pause/resume support */
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this->has_pause_resume = ( snd_pcm_hw_params_can_pause (params)
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&& snd_pcm_hw_params_can_resume (params) );
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printf( "audio_alsa_out:open pause_resume=%d\n", this->has_pause_resume);
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this->sample_rate_factor = (double) this->output_sample_rate / (double) this->input_sample_rate;
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this->bytes_per_frame = snd_pcm_frames_to_bytes (this->audio_fd, 1);
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/*
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* audio buffer size handling
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*/
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/* Copy current parameters into swparams */
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err = snd_pcm_sw_params_current(this->audio_fd, swparams);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to determine current swparams: %s\n", snd_strerror(err));
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goto close;
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}
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/* align all transfers to 1 sample */
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err = snd_pcm_sw_params_set_xfer_align(this->audio_fd, swparams, 1);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set transfer alignment: %s\n", snd_strerror(err));
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goto close;
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}
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/* allow the transfer when at least period_size samples can be processed */
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err = snd_pcm_sw_params_set_avail_min(this->audio_fd, swparams, this->period_size);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set available min: %s\n", snd_strerror(err));
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goto close;
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}
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if (direction == SND_PCM_STREAM_PLAYBACK) {
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/* start the transfer when the buffer contains at least period_size samples */
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err = snd_pcm_sw_params_set_start_threshold(this->audio_fd, swparams, 0);
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} else {
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err = snd_pcm_sw_params_set_start_threshold(this->audio_fd, swparams, -1);
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}
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set start threshold: %s\n", snd_strerror(err));
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goto close;
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}
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if (direction == SND_PCM_STREAM_PLAYBACK) {
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/* never stop the transfer, even on xruns */
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err = snd_pcm_sw_params_set_stop_threshold(this->audio_fd, swparams, 0);
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} else {
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err = snd_pcm_sw_params_set_stop_threshold(this->audio_fd, swparams, this->buffer_size);
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}
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set stop threshold: %s\n", snd_strerror(err));
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goto close;
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}
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/* Install swparams into current parameters */
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err = snd_pcm_sw_params(this->audio_fd, swparams);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set swparams: %s\n", snd_strerror(err));
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goto close;
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}
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#ifdef ALSA_LOG
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snd_pcm_dump_setup(this->audio_fd, jcd_out);
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snd_pcm_sw_params_dump(swparams, jcd_out);
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#endif
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return this->output_sample_rate;
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close:
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snd_pcm_close (this->audio_fd);
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this->audio_fd=NULL;
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return 0;
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}
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int playback_callback(alsa_driver_t *alsa_driver_playback) {
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alsa_driver_t *this = alsa_driver_playback;
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printf("playback callback\n");
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//snd_pcm_writen(this->audio_fd, alsa_buffers, this->period_size);
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}
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int capture_callback(alsa_driver_t *alsa_driver_capture) {
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alsa_driver_t *this = alsa_driver_capture;
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int result;
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#ifdef ALSA_LOG
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printf("capture callback %d samples\n", this->period_size);
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#endif
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snd_pcm_status_t *pcm_stat;
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snd_pcm_status_alloca(&pcm_stat);
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#ifdef ALSA_LOG
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snd_pcm_status(this->audio_fd, pcm_stat);
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snd_pcm_status_dump(pcm_stat, jcd_out);
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#endif
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alsa_buffers[0]=this->app_buffer_y1 + *(this->app_buffer_offset);
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alsa_buffers[1]=this->app_buffer_y2 + *(this->app_buffer_offset);
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result = snd_pcm_readn(this->audio_fd, alsa_buffers, this->period_size);
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*(this->app_buffer_offset) += this->period_size;
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if ( *this->app_buffer_offset >= this->app_buffer_length )
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this->app_buffer_length=0; /* FIXME: implement proper wrapping */
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#ifdef ALSA_LOG
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printf("result=%d\n",result);
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snd_pcm_status(this->audio_fd, pcm_stat);
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snd_pcm_status_dump(pcm_stat, jcd_out);
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#endif
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}
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int capture_xrun(alsa_driver_t *alsa_driver_capture) {
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alsa_driver_t *this = alsa_driver_capture;
|
||||
snd_pcm_status_t *pcm_stat;
|
||||
snd_pcm_status_alloca(&pcm_stat);
|
||||
printf("capture xrun\n");
|
||||
snd_pcm_status(this->audio_fd, pcm_stat);
|
||||
snd_pcm_status_dump(pcm_stat, jcd_out);
|
||||
}
|
||||
|
||||
void ao_alsa_loop(void *iarg) {
|
||||
int playback_nfds;
|
||||
int capture_nfds;
|
||||
struct pollfd *pfd;
|
||||
int nfds;
|
||||
int capture_index;
|
||||
unsigned short playback_revents;
|
||||
unsigned short capture_revents;
|
||||
playback_nfds = snd_pcm_poll_descriptors_count (
|
||||
alsa_driver_playback.audio_fd);
|
||||
capture_nfds = snd_pcm_poll_descriptors_count (
|
||||
alsa_driver_capture.audio_fd);
|
||||
pfd = (struct pollfd *) malloc (sizeof (struct pollfd) *
|
||||
(playback_nfds + capture_nfds));
|
||||
|
||||
nfds=0;
|
||||
#if 0
|
||||
snd_pcm_poll_descriptors (alsa_driver_playback.audio_fd,
|
||||
&pfd[0],
|
||||
playback_nfds);
|
||||
nfds += playback_nfds;
|
||||
#endif
|
||||
snd_pcm_poll_descriptors (alsa_driver_capture.audio_fd,
|
||||
&pfd[nfds],
|
||||
capture_nfds);
|
||||
capture_index = nfds;
|
||||
nfds += capture_nfds;
|
||||
while(1) {
|
||||
if (poll (pfd, nfds, 100000) < 0) {
|
||||
printf("poll failed\n");
|
||||
return;
|
||||
}
|
||||
//snd_pcm_poll_descriptors_revents(alsa_driver_playback.audio_fd, &pfd[0], playback_nfds, &playback_revents);
|
||||
snd_pcm_poll_descriptors_revents(alsa_driver_capture.audio_fd, &pfd[capture_index], capture_nfds, &capture_revents);
|
||||
//if ((playback_revents & POLLERR) || ((capture_revents) & POLLERR)) {
|
||||
if (((capture_revents) & POLLERR)) {
|
||||
printf("pollerr\n");
|
||||
capture_xrun(&alsa_driver_capture);
|
||||
return;
|
||||
}
|
||||
#if 0
|
||||
if (playback_revents & POLLOUT) {
|
||||
playback_callback(&alsa_driver_playback);
|
||||
}
|
||||
#endif
|
||||
if (capture_revents & POLLIN) {
|
||||
capture_callback(&alsa_driver_capture);
|
||||
}
|
||||
}
|
||||
|
||||
return;
|
||||
}
|
||||
|
||||
|
||||
extern void decode1_(int *iarg);
|
||||
void start_threads_(void)
|
||||
int start_threads_(int *ndevin, int *ndevout, short y1[], short y2[],
|
||||
int *nbuflen, int *iwrite, short iwave[],
|
||||
int *nwave, int *nfsample, int *nsamperbuf,
|
||||
int *TRPeriod, int *TxOK, int *ndebug,
|
||||
int *Transmitting, double *Tsec, int *ngo, int *nmode,
|
||||
double tbuf[], int *ibuf, int *ndsec)
|
||||
{
|
||||
pthread_t thread1,thread2;
|
||||
int iret1,iret2;
|
||||
int iarg1=1,iarg2=2;
|
||||
//int32_t rate=11025;
|
||||
int32_t rate=*nfsample;
|
||||
alsa_driver_capture.app_buffer_y1=y1;
|
||||
alsa_driver_capture.app_buffer_y2=y2;
|
||||
alsa_driver_capture.app_buffer_offset=iwrite;
|
||||
alsa_driver_capture.app_buffer_length=nsamperbuf;
|
||||
|
||||
// iret1 = pthread_create(&thread1,NULL,a2d_,&iarg1);
|
||||
iret2 = pthread_create(&thread2,NULL,decode1_,&iarg2);
|
||||
printf("start threads called\n");
|
||||
iret1 = pthread_create(&thread1,NULL,decode1_,&iarg1);
|
||||
/* Open audio card. */
|
||||
ao_alsa_open(&alsa_driver_playback, &rate, SND_PCM_STREAM_PLAYBACK);
|
||||
ao_alsa_open(&alsa_driver_capture, &rate, SND_PCM_STREAM_CAPTURE);
|
||||
|
||||
/*
|
||||
* Start audio io thread
|
||||
*/
|
||||
iret2 = pthread_create(&thread2, NULL, ao_alsa_loop, NULL);
|
||||
snd_pcm_prepare(alsa_driver_capture.audio_fd);
|
||||
snd_pcm_start(alsa_driver_capture.audio_fd);
|
||||
|
||||
/* snd_pcm_start */
|
||||
//iret2 = pthread_create(&thread2,NULL,a2d_,&iarg2);
|
||||
|
||||
}
|
||||
|
||||
14
wrapkarn.c
14
wrapkarn.c
@ -67,3 +67,17 @@ void rs_decode_(int *recd0, int *era0, int *numera0, int *decoded, int *nerr)
|
||||
*nerr=decode_rs_int(rs,recd,era_pos,numera);
|
||||
for(i=0; i<12; i++) decoded[i]=recd[11-i];
|
||||
}
|
||||
|
||||
|
||||
#ifndef WIN32
|
||||
void rs_encode__(int *dgen, int *sent)
|
||||
{
|
||||
rs_encode_(dgen, sent);
|
||||
}
|
||||
|
||||
void rs_decode__(int *recd0, int *era0, int *numera0, int *decoded, int *nerr)
|
||||
{
|
||||
rs_decode_(recd0, era0, numera0, decoded, nerr);
|
||||
}
|
||||
#endif
|
||||
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user